search for: provider2

Hello, I am new on asterisk and do some tests on freepbx. I have 2 SIP provider: Provider1: In-/Out- Flatrate, only 1 Number Provider2: Incoming Flatrate, Outgoing Cost depend on destination, 3 numbers On Asterisk site i have 3 phones (branch ??, don't know how its called in asterisk) Is it possible to do something like: Phone 1: Incoming Call: Number1/Provider1 Outgoing Call: Number1/Provider1 Phone 2: Incoming Call: Numbe...

...risk-users] Rewrite Outgoing Number > Sent by: asterisk-users-bounces at lists.digium.com > > On 14.12.2017 16:30, basti wrote: > Hello, > I am new on asterisk and do some tests on freepbx. > > I have 2 SIP provider: > > Provider1: In-/Out- Flatrate, only 1 Number > Provider2: Incoming Flatrate, Outgoing Cost depend on destination, 3 numbers > > On Asterisk site i have 3 phones > (branch ??, don't know how its called in asterisk) > > Is it possible to do something like: > > Phone 1: Incoming Call: Number1/Provider1 Outgoing Call: Number1/Pr...

....255.0 U 0 0 0 eth0 0.0.0.0 100.10.10.254 0.0.0.0 UG 0 0 0 eth1 Then I've setup two additional routing tables with iproute2 and linked to the main rule list like this # cat >> /etc/iproute2/rt_tables << EOF 10 provider1 20 provider2 EOF # ip route add 100.10.10.0/24 dev eth1 table provider1 # ip route add default via 100.10.10.254 dev eth1 table provider1 # ip rule add from 100.10.10.0/24 table provider1 # ip route add 99.10.11.0/24 dev eth2 table provider2 # ip route add default via 99.10.11.254 dev eth2 table provider2 # i...

...he SDP part I found something like this: o=- 2268929 0 IN IP4 sip2.provider1.de c=IN IP4 sip2.provider1.de If I send Dial(SIP/number|10|m(number)) I have silence on the line. No ringtone, nothing. Now contacting a friend whose Asterisk is connected to another provider (let's give him domain provider2) traced this: Via: SIP/2.0 UDP sip1.provider2.de and its SDP looks like this: o=- 2096205915 2096205915 IN IP4 sip1.provider2.de c=IN IP4 sip1.provider2.de and his early media works fine which means Dialing like the dial above works. The caller can listen to music :) Btw: I wrote hostnames beca...

...re from a VoIP provider they are in e.164 format already, where they're from BT ISDN lines they're usually the last 6 digits of the number. For outbound calls, there's a context called [outboundpstn] containing the following entries: exten => _00.,1,Macro(outbound,{EXTEN},,provider1,provider2,pstn) exten => _0[12]XXXXXXXXX,1,Macro(outbound,${EXTEN},,provider1,provider2,pstn) exten => _07XXXXXXXXX,1,Macro(outbound,${EXTEN},,provider1,provider2,pstn) exten => _08[47]XXXXXX.,1,Macro(outbound,${EXTEN},,provider1,provider2,pstn) exten => _0[58]0XXXXXX.,1,Macro(outbound,${EXTEN},,...

...lace in 1.4.8 but for some reason, the following in my dial plan: ############################################################# exten => _1NXXNXXXXXX,1,GotoIf([${EXTEN} = "15554441212"]?100) exten => _1NXXNXXXXXX,n,Dial(SIP/provider1/${EXTEN},60) exten => _1NXXNXXXXXX,n,Dial(SIP/provider2/${EXTEN},60) exten => _1NXXNXXXXXX,n,Hangup exten => _1NXXNXXXXXX,100,NoOp(Calling my cell w/special CID) exten => _1NXXNXXXXXX,n,Set(CALLERID(all)="Dude" <5551112233>) exten => _1NXXNXXXXXX,n,Dial(SIP/provider1/${EXTEN},60) exten => _1NXXNXXXXXX,n,Dial(SIP/provider2/...

...nsions.conf, there are a number of contexts defined for each group of users, along the lines of: [groupa] [groupb] etc. In each of those, there's a command include => outbound [outbound] has entries similar to the following: exten => _0[123]XXXXXXXX,1,Macro(outbound,${EXTEN}, provider1, provider2) the macro "outbound" is defined in extensions.ael as follows: macro outbound (number, route1, route2) { dosomestuff; } This has worked fine in 1.2 and 1.4, but seems to be choking on 1.6. I've looked through the various changes.txt files, and have read mention of replacing macro c...

...mf=inband. Then the call is transferred to Asterisk 2: RetryDial(/var/lib/asterisk/sounds/connecting,15,10,SIP/12351 at 65.xx.xx.10,,t T,) Where 12351 accepts the call on Asterisk 2, and in some cases, that call is transferred out to a PSTN number, or wherever, but not within Asterisk anymore via provider2, dtmf=rfc2833. When the call comes in, I'd like it to relay DTMF just dandy. How can I do so? There is no NAT between the Asterisk servers or in front of them. However, Asterisk2 has iptables which allows all UDP traffic to/fro Asterisk1. When Asterisk2 transfers the call to external endp...

...ms which I wondered if anyone would be able to help with. Firstly, I wanted to send 0800 calls through 1 sip provider and other 08xx calls through another. I have this: exten => _0800.,1,Dial(SIP/${EXTEN}@provider1,30) exten => _0800.,2,Congestion exten => _08.,1,Dial(SIP/${EXTEN}@provider2,30) exten => _08.,2,Congestion However, whichever way round I put these, 0800 calls still seem to go out of provider2. I fixed this as follows: exten => _08[1-9].,1,Dial(SIP/${EXTEN}@provider2,30) exten => _08[1-9].,2,Congestion ..but, is there another way of getting this working...

Hi, I put on sip.conf the following line #include "sip.d/*.conf" inside I have files like that provider1.conf provider2.conf But asterisk does not want to load it This is the error Sep 1 13:18:35 VERBOSE[8756]: == Parsing '/etc/asterisk/sip.d/*.conf': Sep 1 13:18:35 VERBOSE[8756]: == Parsing '/etc/asterisk/sip.d/*.conf': Not found (No such file or directory) this is the ls result [root@server01...

...(once more building on the example in the section on split-access): ip route add default scope global nexthop via $P1 dev $IF1 weight 1 \ nexthop via $P2 dev $IF2 weight 1 "" What does it mean? It will split the traffic? Eg. some traffic goes through provider1 and some through provider2 ?? Or if provider1 goes down then the packet will go through provider2? I am asking it because I want to create an alternate default gateway when my ADSL link had down. Do you suggest any other solution? Thanks! Leonardo Borda Netwall Tecnologia e Projetos - http://www.netwall.com.br Fone/Fax...

Ok ive been trying to get this to work for about half a year now. Ive searched all over the internet for a solution for my problem. Ive found some solutions, but they only led me to yet more problems. What we want to do is the following: I live in a student complex with 7 other people. Every room has its own internet connection from the same ISP. Ip, gateway, subnet are asigned through dhcp on

...ddress when proxied through openser. Maybe I'm approaching this from the wrong direction, anyone have any ideas? Mike [privider1a] type=peer host=67.x.x.x insecure=invite,port context=default qualify=999 [provider1a] type=peer host=67.x.x.x insecure=invite,port context=default qualify=999 [provider2] type=peer ;host=sip.provider2.com host=64.x.x.x insecure=invite,port context=default qualify=999

Hi, How can I setup Asterisk to place calls if the same dial pattern can be routed through several PRI gateways. I have one way that I tried: exten => _9737XXXX,1,Dial(SIP/${EXTEN:1}@172.17.99.5) exten => _9737XXXX,2,Dial(SIP/${EXTEN:1}@172.17.99.6) exten => _9737XXXX,3,Dial(SIP/${EXTEN:1}@172.17.99.7) exten => _9737XXXX,4,Congestion exten => _9737XXXX,102,Busy

Hello, I have the following setup: sip phones <->SER <-> asterisk <-> voip provider1 <-> voip provider2 i got a toll-free DID from voipprovider1 to allow people from outside to call into asterisk, get authenticated, and use voipprovider2 to call out (kind of a primitive calling card app). anyway, voiprovider is giving my bad DTMF (usually 2 keypresses for every 1)...transport is via SIP, i am regist...

Is there a way to override the fromdomain specified in the sip.conf and instead set the value from the dialplan? If we use: Set(CALLERID(num)=user at domain.com The SIP From header turns into: user at domain.com@10.10.10.10 We want user at domain.com, and we can't have an entry in sip.conf for every provider. -- Eric Chamberlain

Hi I want to be able to use one provider if I dial 0 before the number and another if I dial 1 before, how can I do that in asterisk 1.6? /ralf Ralf Tr?skman, IT AdLibris AB, Box 3667, 103 59 Stockholm. Bes?ksadress: Sveav?gen 56C, 111 34, Stockholm - Obs ny address! Dir: +46-(0)8-5460 60 91, mob: +46-(0)70-7548074, vxl: +46-(0)8-5460 60 00, fax: +46-(0)8-5460 60 99 ralf at

...to the wrong interface. I''m not sure to understand rules and routes traversal correctly (I couldn''t find answers in the howto). So, here are my questions: 1. How does the rule traversal work exactly? If I have rules like this: 100: from all lookup provider1 200: from all lookup provider2 What does make stop the traversal? For instance, does a default route in provider1 would stop the traversal? 2. are tables entirely separated? Say, if you have this: 100: from all lookup table_part1 200: from all lookup table_part2 and table_part1 contains the route for a provider and table_part2...

...ne modem each. AlexRouter and DaveRouter. They run Bering-uClibc 2.x off of fd0. A wired/wireless network connects the two together. 192.168.58.0/24. AlexRouter is the default route/DNS server/DHCP server for every host on the network. It gets its DNS servers from dhcpcd. They way I figure it, Provider2 in the example is (in my case) actually DaveRouter. With that in mind, these are the figures I came up with for settings up the routes. These are all from the perspective of AlexRouter. $IF1 = eth0 $IF2 = br0 $IP1 = 80.blah.blah.blah (can''t remember my real address) $IP2 = 192.168.58.1 $P...

...ne modem each. AlexRouter and DaveRouter. They run Bering-uClibc 2.x off of fd0. A wired/wireless network connects the two together. 192.168.58.0/24. AlexRouter is the default route/DNS server/DHCP server for every host on the network. It gets its DNS servers from dhcpcd. They way I figure it, Provider2 in the example is (in my case) actually DaveRouter. With that in mind, these are the figures I came up with for settings up the routes. These are all from the perspective of AlexRouter. $IF1 = eth0 $IF2 = br0 $IP1 = 80.blah.blah.blah (can''t remember my real address) $IP2 = 192.168.58.1 $P...